similar to: detecting pickup

Hi! I am playing around with Asterisk and have a problem :-) (Asterisk-version: 1.2.4, chan_capi-cm-version: 0.6.4) I have a sip-phone at my desk and an ISDN-phone (independent of the Asterisk-server) in my living room, when I'm not at my desk, the sip-phone is switched off. I would like to be able to accept calls at both phones (when available) and have Voicemail kick in if I don't

I'm working through my list of features people will expect, and Hold Pickup is at the top at the moment -- has anyone done any work on this? We've had some unpleasant experiences with call parking, and everyone seems to like the Hold Pickup model. If you don't know what I mean by Hold Pickup, it's sort of a reverse transfer; pick up the nearest phone and dial

I have a small problem, Whenever we pickup a call using *8 then try to transfer it via flash or # transfer the call is dropped. Any ideas? Whe have all called exten's in extension.conf ending in |t som that # transfers work. What am I doing wrong? Chad Sawyer, Manager, Network Administrator -------------- next part -------------- An HTML attachment was scrubbed... URL:

<FONT face="Default Sans Serif, Verdana, Arial, Helvetica, sans-serif" size=2><DIV>I have been playing with getting the sample.call file to work by dropping it into /var/spool/asterisk/outgoing.&nbsp; The process works to the point of calling the desired number and plays the message.&nbsp; The problem is that the message starts playing almost immediately, so if the

I am using asterisk at home with a Cisco ATA186 and a clone X100P card. My inbound telco line is plugged into the X100P card. My telco line is also plugged into other phones in the house for now so someone else can answer the phone without asterisk being involved. What I would like to do is if someone has answered the call on a normal phone in the house I would like to be able to join the call

Hi, I have asterisk connected to PSTN via H.323 gateway via chan_oh323. Incoming calls to SIP extensions work, but SIP message "486 busy here" from a busy extension isn't correctly forwarded to H.323. As a result, a caller from the H.323 side calling a busy SIP extension gets some rings and then an irritating timeout with H.323 message 'no user responding' instead of

Hi guys, I spent a couple of hours in Google, but the problem appears to be uncommon, so I'd like to ask about it here. The problem is exactly the opposite to "Asterisk does not detect FXO hangup". In my case it's the Telco who does not appear to be detecting Asterisk's hangups. Telco is Telus in Vancouver, Canada. The setup is very simple - Telco -> FXO/TDM400p

We have 4 analog line and 2 analog trunks. On the trunks we have all the DIDs coming into the current phone system. Trying to get everything moved over to Asterisk but having issues picking up the calls on the analog trunk. We can receive calls on the plain analog lines and we can call out on all analog lines and analog trunks. When a call comes in on the trunk line the ZAP channels don't

The problem: T100P connected to an Adit600. Channel 1-16 are FXS, 17-24 FXO. I have Zap/24 in callgroup 3 and Zap/1-16 in pickupgroup 3. When a call comes in on Zap/24 I cannot pick it up with *8 from Zap/1-16. *CLI> show version Asterisk CVS-04/27/04-23:48:08 built by root@tuck on a i686 running Linux The zapata.conf and extensions.conf are located here:

Hi, I've used various patches with asterisk 1.4 to have support for call pickup and notification with good results. Now I'm trying vanilla 1.6.2 with its official support for "dialog-info +xml" notifications with no success. This is what i'm doing: - Phone A has a key configured as type "extension" pointing to Phone B. - In sip.conf I added

Hello asterisk-users, My SIP phones are in pickupgroup, and if some of them ringing from other phone can pick up with *8 as usual. But I want to know if this happen. I've tried the a extension, but seems not working. Any other idea? -- Best regards, Gergo mailto:csibra at gmail.com

Hi All, I'm using Asterisk 1.2.14 under openSuSE 10.2 with kernel 2.6.18. I have Wildcard TDM400P card and D-Link DPH-120S and DPH-140S SIP phones. I would like to be able to pickup ringing extention from any SIP phone using Pickup() application. from my dial plan: [incoming] exten => s,1,Dial(SIP/somebody1|60|tTrR) [internal] include => outbound-local include => parkedcalls

Please help! When I try to place a call pickup from a cisco phone 7960 using *8 the call is picked up but the other phone continues ringing. Is there any problem with call pickup in SIP.

Hi everyone: The following code dials our prefix, sends a beep, and sends a DTMF "c" tone, then dials the phone number. I need to send the DTMF only if the phone is answered. [voip] exten=>i,1,NoCDR() exten=>i,2,Hangup() exten=>s,1,Wait(2) exten=>s,2,Background(beep||) exten=>s,3,DigitTimeout(6) exten=>s,4,ResponseTimeout(10) exten=>s,5,SendDTMF(c)

Hello users, I have been working with a conference setup. My setup includes: 1)There will be an interface number provided to the user which might be a DID number or A Toll free number When user calls the number it asks for the conference room number and the user pin . on successfull authentication he will be participated in the conference 2)by didaling the same DID number the

I must be getting something wrong about this call pickup. In zapata.conf I have just the default callgroup=1 and pickupgroup=1. If I call from my mobile to * and then try to dial *8 from any other phone than the one which is ringing I just get a "Nothing to pick up" answer on my * console. I also have experimented with those parameters in sip.conf but are not aware of exactly where to

Hi everybody, I'm facing a strange problem after upgrading Asterisk from 1.0.9 to 1.2.4. Sometimes, when receiving an incoming call from pstn, although my sip phones ring correctly (I've got both softphones and hardware phones), noone can pick up the call. Asterisk CLI shows me that the phones are ringing, then nothing happens, so there's no problem _after_ someone picked up, simply

Hello, We are faced with a problem concerning queues. When we have several calls in different queues, is there some sort of way to open a channel between a (sip-)phone and a SPECIFIC call in a queue using the Asterisk manager api? We would like to do this even when we are not a member of that specific queue. Thanks in advance for any suggestions! cheers

Hi, What do I need to do to get asterisk to NOT pickup a Zap channel when it rings? The channel in question is used for outbound calls only, and all incoming calls are answered by an analog phone elsewhere in the building that does not run through asterisk... so.. either make it not answer.. or make it delay for like 90 seconds.. I've tried wait's.. but it still seems to pickup the

Can the Manager Redirect command transfer a ringing SIP extension? I'm trying to implement a Camp On feature, and having failed to do it in Dial Plan logic, am trying to do it with manager logic. If an arbitrary Sip extension is ringing, I need the ability to pick up that extension from any other phone. What little docs there are on Manager commands shows Redirect takes these parameters: